/external/chromium_org/third_party/libjingle/source/talk/session/media/ |
mediarecorder.h | 85 talk_base::StreamInterface* send_stream, 89 talk_base::StreamInterface* send_stream, 107 talk_base::StreamInterface* send_stream,
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mediarecorder.cc | 122 talk_base::StreamInterface* send_stream, 125 return InternalAddChannel(channel, false, send_stream, recv_stream, 129 talk_base::StreamInterface* send_stream, 132 return InternalAddChannel(channel, true, send_stream, recv_stream, 138 talk_base::StreamInterface* send_stream, 153 sink_pair->send_sink.reset(new RtpDumpSink(send_stream));
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/external/chromium_org/third_party/webrtc/video/ |
loopback.cc | 90 VideoSendStream* send_stream = local 96 test::VideoCapturer::Create(send_stream->Input(), 114 send_stream->Start(); 120 send_stream->Stop(); 124 call->DestroyVideoSendStream(send_stream);
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call_perf_tests.cc | 312 VideoSendStream* send_stream = local 317 test::FrameGeneratorCapturer::Create(send_stream->Input(), 323 send_stream->Start(); 340 send_stream->Stop(); 350 sender_call->DestroyVideoSendStream(send_stream); 471 VideoSendStream* send_stream = local 474 test::FrameGeneratorCapturer::Create(send_stream->Input(), 504 send_stream->Start(); 512 send_stream->Stop(); 516 sender_call->DestroyVideoSendStream(send_stream); 665 VideoSendStream* send_stream = local [all...] |
call.cc | 77 virtual void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) 189 VideoSendStream* send_stream = new VideoSendStream( local 203 send_ssrcs_[config.rtp.ssrcs[i]] = send_stream; 205 return send_stream; 208 void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) { 209 assert(send_stream != NULL); 218 if (it->second == static_cast<VideoSendStream*>(send_stream)) {
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rampup_tests.cc | 253 virtual void SetSendStream(const VideoSendStream* send_stream) { 255 send_stream_ = send_stream; 490 VideoSendStream* send_stream = local 494 test::FrameGeneratorCapturer::Create(send_stream->Input(), 500 send_stream->Start(); 506 send_stream->Stop(); 508 call->DestroyVideoSendStream(send_stream); 543 VideoSendStream* send_stream = local 545 stream_observer.SetSendStream(send_stream); 559 test::FrameGeneratorCapturer::Create(send_stream->Input() [all...] |
full_stack.cc | 412 VideoSendStream* send_stream = local 414 analyzer.input_ = send_stream->Input(); 440 send_stream->Start(); 446 send_stream->Stop(); 450 call->DestroyVideoSendStream(send_stream);
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video_send_stream_tests.cc | [all...] |
/external/chromium_org/third_party/webrtc/ |
call.h | 96 virtual void DestroyVideoSendStream(VideoSendStream* send_stream) = 0;
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/external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
webrtcvideoengine2_unittest.cc | 206 void FakeCall::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) { 208 static_cast<FakeVideoSendStream*>(send_stream); 635 FakeVideoSendStream* send_stream = local 639 ASSERT_EQ(1u, send_stream->GetConfig().rtp.extensions.size()); 640 EXPECT_EQ(id, send_stream->GetConfig().rtp.extensions[0].id); 641 EXPECT_EQ(webrtc_ext, send_stream->GetConfig().rtp.extensions[0].name); 654 EXPECT_FALSE(send_stream->GetConfig().rtp.extensions.empty()); 774 FakeVideoSendStream* send_stream = AddSendStream( local 777 ASSERT_EQ(rtx_ssrcs.size(), send_stream->GetConfig().rtp.rtx.ssrcs.size()); 779 EXPECT_EQ(rtx_ssrcs[i], send_stream->GetConfig().rtp.rtx.ssrcs[i]) 813 FakeVideoSendStream* send_stream = local 874 FakeVideoSendStream* send_stream = local 894 FakeVideoSendStream* send_stream = local [all...] |
webrtcvideoengine2_unittest.h | 109 webrtc::VideoSendStream* send_stream) OVERRIDE;
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